The radio frequency spectrum associated with FM broadcasting band ranges from 88 to 107 MHz, and subdivided into channels. There is a portion of bandwidth within each FM channel that is not required for transmitting the main FM station broadcast signal, and is generally under utilized. Those responsible for the FM stations have continuously tried to efficiently utilize all of their allotted frequency resource including this available excess bandwidth.
Under current Federal Communications Commission (FCC) rules, FM radio stations are each allocated a channel that is 200 kHz wide. United States FM radio stations are granted a license to operate an FM radio signal within an assigned range of frequencies called a channel. This range is substantially larger than the minimum range or bandwidth required for the main FM radio signal. Although a typical FM station is assigned a bandwidth of 200 KHz, the positive or one-sided baseband frequency spectrum is a 100 KHz bandwidth, and an FM station takes up to a maximum of 53 KHz for the main FM stereo broadcasting station, and less for a monoaural station. The remaining portion of the baseband signal from 53 KHz to 99 KHz, approximately 50% of the available FM channel spectrum resource, is not required for broadcasting the main FM station signal.
The present FM radio stations with an operating bandwidth of 100 kHz contains two audio channels. The main channel, known as FM stereo, is transmitted using an L+R and L−R technique to accommodate monoaural reception as well. This is the channel received on car radios and similar mediums. A secondary channel, or subcarrier, above the 53 KHz area is used for Muzak and other private audio broadcasts. In general, the subcarrier is an under utilized bandwidth.
A diagrammatic representation of an FM channel is shown in FIG. 1 showing the 100 kHz one-sided baseband frequency. For the signals operating between 88–107 MHz, there is a main channel from 0–15 kHz, which is the mono band (L+R). There is a 19 KHz stereo pilot signal and the stereo band is from 23–53 kHz, wherein the lower side band (LSB) spans 23–38 kHz, the upper side band (USB) spans from 38 kHz to 53 kHz, and the (L−R) is at 38 kHz. The subcarrier band extends from 53 kHz to 99 kHz.
Experiments in the FM subcarrier bandwidth has been ongoing for many years, used in a number of applications, and implemented in a variety of analog and digital communication schemes. For example, Muzak, is standardized music used in elevators and similar locals. Muzak uses a double side band AM modulation of a 67 KHz subcarrier to carry the subscription music.
Radio stations try to lease frequencies in the “excess” bandwidth to other users through various subcarrier based systems. One such service is known as Sub-Carrier administration (SCA), that has been used in the United States for many years for background music without commercial interruption, reading services for the blind, stock market information, and educational and religious applications, and paging services to name a few. U.S. Pat. No. 5,248,610 shows the subcarrier band employed for transmission of traffic information, while a paging scheme that employs the subcarrier band is disclosed in U.S. Pat. No. 6,088,577 to transmit voice pager data over the FM subcarrier band.
SCA has also been used for data transmission, having the ability to reliably support a data rate of 4,800 bits/second or higher. The FCC deregulated SCA service and stations are free-to carry SCA services without prior authorization, as long as all uses of the frequency are within the regulations imposed on the license holder.
In Europe, one FM subcarrier application is known as the Radio Data System (RDS) while in the U.S., it is known as the Radio Broadcast Data Service (RDBS). The RDS system has been implemented by the BBC on the BBC FM transmissions in England. Similar systems are available in several European countries, wherein these systems use an RDS subcarrier at 57 kHz that is modulated with data signals. The RDS system has a number of message group types. In these applications, a 57 KHz subcarrier is modulated using bi-polar phase shift keying (BPSK) to carry a low speed digital data signal. A block and bit synchronization method as well as a simple linear block encoding for error detection and correction enable the system to function. The data rate for the RDS system is 1187.5 bits/second or approximately 11.4 groups/second, and the channel modulation efficiency of RDS is about 0.3 bps/Hz. RDS provides for additional functional features such as roaming, seeking and locking onto of FM stations transmitting RDS signals.
Since systems like the RDS system have broad coverage, a number of users can use a data channel on a pro-rata basis. RDS is a fairly robust digital subcarrier communication scheme because of the long baud interval, low subcarrier frequency, and narrow bandwidth. This schema has been adopted as an international standard and incorporates specification of the physical layer (the modulation and FM interface), the data link layer (error correction coding), and a network layer for service delivery.
RDS has been popular in European countries for transmitting traffic-related information to motorists while utilizing the existing FM radio broadcasting infrastructure. However, RDS has a slow data transmission rate and with the many function groups RDS is too slow for effective data transmission.
Due to the low data rate of RDS, another format called the Data Radio Channel (DARC) supports a higher data rate FM subcarrier service. DARC is also an international standards (EIA-794) and has several different modes of operation. The modes vary according to the amount of error correction coding (ECC) overhead applied to the data transmission. DARC is 16K bits per second minimum-shift keyed modulation of a 76K Hz subcarrier tone.
Another higher frequency system that is comparable to DARC is the Subcarrier Traffic Information Channel (STIC). This digital system uses a differentially encoded, quadrature phase-shift keyed modulation of either a 72.2K Hz of a 87.4K Hz subcarrier tone to deliver a 18,050 or 21,850 bps raw data rate. STIC also has a US standard (EIA-795) and also addresses the STIC standard addresses layers 1 through 4. STIC basically applies modern modem technology to a FM subcarrier system by using efficient convolutional coding, code concatenation and interleaving at the bit level to address channel impairments.
Several other higher speed subcarrier technologies have been developed with limited success, demonstrating the difficulty of the propagation environment to which FM subcarrier systems are subjected and the rather low efficiency of the current FM subcarrier systems.
In typical FM broadcasting, left and right stereo base band signals are low-pass filtered and combined to produce a composite stereo signal. The circuit that combines the left and right component signals and produces the composite stereo signal is called an exciter. The composite stereo signal is used to drive a FM modulator that modulates a carrier wave in accordance with the composite signal. The modulated carrier wave is then broadcast using a FM antenna as an analog signal.
Conventional systems generate the composite stereo signal using analog equipment. But, there are a many difficulties in generating the composite stereo signal in the analog format. For example, low-pass filtering and sub-carrier stereo modulation are complicated for an analog system and the analog filters introduce phase distortions and group delay distortions into the resulting signal.
There have been attempts at generating the stereo composite signal in a digital format and converting the signal to an analog signal for broadcasting. This is advantageous for radio stations as employ the digital methods of programming such as CD's, DAT and computer audio files. Using digital signal processing (DSP) with fast high precision A/D and D/A converters, a FM exciter using digital signal processing has better performance than the analog systems.
And, there have been attempts at generating the audio signal in a digital format, wherein the signal is broadcast in digital, termed Digital Audio Broadcasting (DAB). While producing superior audio quality, the radio receivers must be equipped to receive the DAB. The DAB is more comparable to streaming audio over the Internet than the conventional analog broadcasts. Radio receivers buffer the incoming signals and play back the buffered elements. In the US, there are several proposed systems including the In-Band On-Channel (IBOC) and the In-Band Adjacent Channel (IBAC). Most other countries are backing the Eureka 147 standard. The Eureka 147 system employs similar attributes but occupies a different section of the radio spectrum. More information about the IBOC methodology is described in U.S. Pat. No. 5,949,796.
Unfortunately, pure DAB systems will require every person to purchase a new radio receiver as the analog receivers do not process the digital audio signals. There are proposals for hybrid processing using both analog and digital broadcasting for a period to allow the consumers to transition to digital receivers.
While DAB standards have not yet been finalized and there is still uncertainty in the implementation, the benefits in quality, efficiency, ease and cost will eventually make some form of DAB a reality. And, with lower cost broadcast systems there will likely be an increased number of applications in the subcarrier band for various transmissions.
One implementation for a digital signal processing system of a digital FM exciter provides for the left channel to provide a left analog audio signal that becomes the left component of the composite stereo signal. Similarly, the right channel provides a right analog audio signal that becomes the right component of the composite stereo signal. The left and right analog signals are respectively processed by anti-aliasing filters. After filtering, the left and right signals are respectively converted from analog into digital signals by A/D converters. The converted digital signals are provided to a digital signal processor (DSP).
The DSP combines the left and right signals into a composite digital signal. More specifically, the DSP performs band limiting filtering, pre-emphasizing, left and right channel mixing, sub-carrier generation, sub-carrier modulation and Sin(x)/x compensation for the D/A converter. Additionally, the DSP provides soft level limiting, loudness signal monitoring for analog and digital automatic gain control, and spectrum analysis for optimized system control and operation.
The composite digital signal output by the DSP is then converted to an analog signal by D/A converter and filtered through low pass filter. The result is a composite analog base-band stereo signal that may be used to modulate a carrier wave which is then broadcast by an FM antenna.
The drawbacks of this system result from the fact that the D/A converter and the external analog FM modulator must be of the highest quality, and therefore are very expensive. The high quality processing achieved by the front end A/D converters and the DSP will be lost if the D/A converter and analog FM modulator cannot match the performance of the DSP.
There are many related patents, including U.S. Pat. No. 6,295,362 involves a digital FM signal generator that generates a modulated FM signal for broadcasting without the need for an analog modulator. There is a FM subcarrier system that multiplexes the Data Radio Channel (DARC) encoded source channels using an FM subcarrier.
As discussed, the use of digital inputs in the broadcasting system improves the efficiency and cost. In particular, the audio inputs can arrive in one of the many digital formats. One of the present popular audio formats is the Moving Picture Experts Group (MPEG) audio layer-3 (MP3) that is a technique for audio compression and is basically a standard way of compressing/decompressing digital multimedia such as music. The MP3 protocol compresses the sound signal into a small-capacity music file at the rate of 1:10 by applying the MP3 technique to be more convenient to save and to speed up the network transmission rate.
As MP3 is a compressed audio file format, it is one of the more popular audio formats on the Internet, due to the small size and near CD quality. MP3 is a method of compressing audio samples with minimal loss of quality, with compression of up to 12:1 having no loss of quality.
There are various MP3 encoders/decoders for converting between audio formats. MP3 provides various levels of quality depending upon the number of bits used for encoding. There are various levels of compression quality supported in the standard such as 96 kb/s, 128 kb/s, 160 kb/s . . . . in 32 kb increments. The higher the transmission rate the better the quality.
What is needed is a system to improve audio fidelity and enhance the quality of music. Such a system should capitalize on new technologies but be implemented within the bounds of the current radio community.